[FFmpeg-devel] [PATCH] Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`.

* [FFmpeg-devel] [PATCH] Hi! We've been fuzzing `ffmpeg` with [sydr-fuzz](https://github.com/ispras/oss-sydr-fuzz) security predicates and we found numeric truncation error in `svs.c:57`.
@ 2023-09-20 12:18 mezhuevtp
  2023-09-20 12:29 ` Paul B Mahol
  2023-09-20 22:51 ` Andreas Rheinhardt
  0 siblings, 2 replies; 4+ messages in thread
From: mezhuevtp @ 2023-09-20 12:18 UTC (permalink / raw)
  To: ffmpeg-devel; +Cc: mezhuevtp, headshog

From: headshog <craaaaaachind@gmail.com>

In function `svs_read_header` on line 57 field `st->codecpar->sample_rate` has type `int`, the type of return value in `av_rescale_rnd` function is `uint64_t`, so the numeric truncation may occur here. Then value of `st->codecpar->sample_rate` is passed to `avpriv_set_pts_info` function parameter `unsgined int pts_den`. In this function `pts_den` is used only in passing its value to parameter `int64_t den` in function `av_reduce`. So we suggest to change the type of field `sample_rate` to `int64_t` and to change the type of `pts_den` to `uint64_t` in `avpriv_set_pts_info` function. The other way to solve this is to add a checker for `sample_rate` valid value.

- OS: ubuntu 20.04
- commit: f225f8d7464569c7b917015c26ad30a37a5fbbe2

```
libavformat/svs.c:57:36: runtime error: implicit conversion from type 'int64_t' (aka 'long') of value 6321554672 (64-bit, signed) to type 'int' changed the value to 2026587376 (32-bit, signed)
SUMMARY: UndefinedBehaviorSanitizer: undefined-behavior libavformat/svs.c:57:36
```
---
 libavcodec/codec_par.h | 2 +-
 libavformat/internal.h | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

diff --git a/libavcodec/codec_par.h b/libavcodec/codec_par.h
index add90fdb1e..2bbb7ba4e7 100644
--- a/libavcodec/codec_par.h
+++ b/libavcodec/codec_par.h
@@ -175,7 +175,7 @@ typedef struct AVCodecParameters {
     /**
      * Audio only. The number of audio samples per second.
      */
-    int      sample_rate;
+    int64_t      sample_rate;
     /**
      * Audio only. The number of bytes per coded audio frame, required by some
      * formats.
diff --git a/libavformat/internal.h b/libavformat/internal.h
index 901a8b51c6..6e1fce41aa 100644
--- a/libavformat/internal.h
+++ b/libavformat/internal.h
@@ -584,7 +584,7 @@ const struct AVCodec *ff_find_decoder(AVFormatContext *s, const AVStream *st,
  * @param pts_den time base denominator
  */
 void avpriv_set_pts_info(AVStream *st, int pts_wrap_bits,
-                         unsigned int pts_num, unsigned int pts_den);
+                         unsigned int pts_num, uint64_t pts_den);
 
 /**
  * Set the timebase for each stream from the corresponding codec timebase and
-- 
2.25.1

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